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Network Working Group S. Casner
Request for Comments: 2508 Cisco Systems
Category: Standards Track V. Jacobson
Cisco Systems
February 1999
Compressing IP/UDP/RTP Headers for Low-Speed Serial Links
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (1999). All Rights Reserved.
Abstract
This document describes a method for compressing the headers of
IP/UDP/RTP datagrams to reduce overhead on low-speed serial links.
In many cases, all three headers can be compressed to 2-4 bytes.
Comments are solicited and should be addressed to the working group
mailing list rem-conf@es.net and/or the author(s).
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119.
1. Introduction
Since the Real-time Transport Protocol was published as an RFC [1],
there has been growing interest in using RTP as one step to achieve
interoperability among different implementations of network
audio/video applications. However, there is also concern that the
12-byte RTP header is too large an overhead for 20-byte payloads when
operating over low speed lines such as dial-up modems at 14.4 or 28.8
kb/s. (Some existing applications operating in this environment use
an application-specific protocol with a header of a few bytes that
has reduced functionality relative to RTP.)
Header size may be reduced through compression techniques as has been
done with great success for TCP [2]. In this case, compression might
be applied to the RTP header alone, on an end-to-end basis, or to the
combination of IP, UDP and RTP headers on a link-by-link basis.
Compressing the 40 bytes of combined headers together provides
substantially more gain than compressing 12 bytes of RTP header alone
because the resulting size is approximately the same (2-4 bytes) in
either case. Compressing on a link-by-link basis also provides
better performance because the delay and loss rate are lower.
Therefore, the method defined here is for combined compression of IP,
UDP and RTP headers on a link-by-link basis.
This document defines a compression scheme that may be used with
IPv4, IPv6 or packets encapsulated with more than one IP header,
though the initial focus is on IPv4. The IP/UDP/RTP compression
defined here is intended to fit within the more general compression
framework specified in [3] for use with both IPv6 and IPv4. That
framework defines TCP and non-TCP as two classes of transport above
IP. This specification creates IP/UDP/RTP as a third class extracted
from the non-TCP class.
2. Assumptions and Tradeoffs
The goal of this compression scheme is to reduce the IP/UDP/RTP
headers to two bytes for most packets in the case where no UDP
checksums are being sent, or four bytes with checksums. It is
motivated primarily by the specific problem of sending audio and
video over 14.4 and 28.8 dialup modems. These links tend to provide
full-duplex communication, so the protocol takes advantage of that
fact, though the protocol may also be used with reduced performance
on simplex links. This compression scheme performs best on local
links with low round-trip-time.
This specification does not address segmentation and preemption of
large packets to reduce the delay across the slow link experienced by
small real-time packets, except to identify in Section 4 some
interactions between segmentation and compression that may occur.
Segmentation schemes may be defined separately and used in
conjunction with the compression defined here.
It should be noted that implementation simplicity is an important
factor to consider in evaluating a compression scheme.
Communications servers may need to support compression over perhaps
as many as 100 dial-up modem lines using a single processor.
Therefore, it may be appropriate to make some simplifications in the
design at the expense of generality, or to produce a flexible design
that is general but can be subsetted for simplicity. Higher
compression gain might be achieved by communicating more complex
models for the changing header fields from the compressor to the
decompressor, but that complexity is deemed unnecessary. The next
sections discuss some of the tradeoffs listed here.
2.1. Simplex vs. Full Duplex
In the absence of other constraints, a compression scheme that worked
over simplex links would be preferred over one that did not.
However, operation over a simplex link requires periodic refreshes
with an uncompressed packet header to restore compression state in
case of error. If an explicit error signal can be returned instead,
the delay to recovery may be shortened substantially. The overhead
in the no-error case is also reduced. To gain these performance
improvements, this specification includes an explicit error
indication sent on the reverse path.
On a simplex link, it would be possible to use a periodic refresh
instead. Whenever the decompressor detected an error in a particular
packet stream, it would simply discard all packets in that stream
until an uncompressed header was received for that stream, and then
resume decompression. The penalty would be the potentially large
number of packets discarded. The periodic refresh method described
in Section 3.3 of [3] applies to IP/UDP/RTP compression on simplex
links or links with high delay as well as to other non-TCP packet
streams.
2.2. Segmentation and Layering
Delay induced by the time required to send a large packet over the
slow link is not a problem for one-way audio, for example, because
the receiver can adapt to the variance in delay. However, for
interactive conversations, minimizing the end-to-end delay is
critical. Segmentation of large, non-real-time packets to allow
small real-time packets to be transmitted between segments can reduce
the delay.
This specification deals only with compression and assumes
segmentation, if included, will be handled as a separate layer. It
would be inappropriate to integrate segmentation and compression in
such a way that the compression could not be used by itself in
situations where segmentation was deemed unnecessary or impractical.
Similarly, one would like to avoid any requirements for a reservation
protocol. The compression scheme can be applied locally on the two
ends of a link independent of any other mechanisms except for the
requirements that the link layer provide some packet type codes, a
packet length indication, and good error detection.
Conversely, separately compressing the IP/UDP and RTP layers loses
too much of the compression gain that is possible by treating them
together. Crossing these protocol layer boundaries is appropriate
because the same function is being applied across all layers.
3. The Compression Algorithm
The compression algorithm defined in this document draws heavily upon
the design of TCP/IP header compression as described in RFC 1144 [2].
Readers are referred to that RFC for more information on the
underlying motivations and general principles of header compression.
3.1. The basic idea
In TCP header compression, the first factor-of-two reduction in data
rate comes from the observation that half of the bytes in the IP and
TCP headers remain constant over the life of the connection. After
sending the uncompressed header once, these fields may be elided from
the compressed headers that follow. The remaining compression comes
from differential coding on the changing fields to reduce their size,
and from eliminating the changing fields entirely for common cases by
calculating the changes from the length of the packet. This length
is indicated by the link-level protocol.
For RTP header compression, some of the same techniques may be
applied. However, the big gain comes from the observation that
although several fields change in every packet, the difference from
packet to packet is often constant and therefore the second-order
difference is zero. By maintaining both the uncompressed header and
the first-order differences in the session state shared between the
compressor and decompressor, all that must be communicated is an
indication that the second-order difference was zero. In that case,
the decompressor can reconstruct the original header without any loss
of information simply by adding the first-order differences to the
saved uncompressed header as each compressed packet is received.
Just as TCP/IP header compression maintains shared state for multiple
simultaneous TCP connections, this IP/UDP/RTP compression SHOULD
maintain state for multiple session contexts. A session context is
defined by the combination of the IP source and destination
addresses, the UDP source and destination ports, and the RTP SSRC
field. A compressor implementation might use a hash function on
these fields to index a table of stored session contexts. The
compressed packet carries a small integer, called the session context
identifier or CID, to indicate in which session context that packet
should be interpreted. The decompressor can use the CID to index its
table of stored session contexts directly.
Because the RTP compression is lossless, it may be applied to any UDP
traffic that benefits from it. Most likely, the only packets that
will benefit are RTP packets, but it is acceptable to use heuristics
to determine whether or not the packet is an RTP packet because no
harm is done if the heuristic gives the wrong answer. This does
require executing the compression algorithm for all UDP packets, or
at least those with even port numbers (see section 3.4).
Most compressor implementations will need to maintain a "negative
cache" of packet streams that have failed to compress as RTP packets
for some number of attempts in order to avoid further attempts.
Failing to compress means that some fields in the potential RTP
header that are expected to remain constant most of the time, such as
the payload type field, keep changing. Even if the other such fields
remain constant, a packet stream with a constantly changing SSRC
field SHOULD be entered in the negative cache to avoid consuming all
of the available session contexts. The negative cache is indexed by
the source and destination IP address and UDP port pairs but not the
RTP SSRC field since the latter may be changing. When RTP
compression fails, the IP and UDP headers MAY still be compressed.
Fragmented IP Packets that are not initial fragments and packets that
are not long enough to contain a complete UDP header MUST NOT be sent
as FULL_HEADER packets. Furthermore, packets that do not
additionally contain at least 12 bytes of UDP data MUST NOT be used
to establish RTP context. If such a packet is sent as a FULL_HEADER
packet, it MAY be followed by COMPRESSED_UDP packets but MUST NOT be
followed by COMPRESSED_RTP packets.
3.2. Header Compression for RTP Data Packets
In the IPv4 header, only the total length, packet ID, and header
check-sum fields will normally change. The total length is redundant
with the length provided by the link layer, and since this
compression scheme must depend upon the link layer to provide good
error detection (e.g., PPP's CRC [4]), the header checksum may also
be elided. This leaves only the packet ID, which, assuming no IP
fragmentation, would not need to be communicated. However, in order
to maintain lossless compression, changes in the packet ID will be
transmitted. The packet ID usually increments by one or a small
number for each packet. (Some systems increment the ID with the
bytes swapped, which results in slightly less compression.) In the
IPv6 base header, there is no packet ID nor header checksum and only
the payload length field changes.
In the UDP header, the length field is redundant with the IP total
length field and the length indicated by the link layer. The UDP
check-sum field will be a constant zero if the source elects not to
generate UDP checksums. Otherwise, the checksum must be communicated
intact in order to preserve the lossless compression. Maintaining
end-to-end error detection for applications that require it is an
important principle.
In the RTP header, the SSRC identifier is constant in a given context
since that is part of what identifies the particular context. For
most packets, only the sequence number and the timestamp will change
from packet to packet. If packets are not lost or misordered
upstream from the compressor, the sequence number will increment by
one for each packet. For audio packets of constant duration, the
timestamp will increment by the number of sample periods conveyed in
each packet. For video, the timestamp will change on the first
packet of each frame, but then stay constant for any additional
packets in the frame. If each video frame occupies only one packet,
but the video frames are generated at a constant rate, then again the
change in the timestamp from frame to frame is constant. Note that
in each of these cases the second-order difference of the sequence
number and timestamp fields is zero, so the next packet header can be
constructed from the previous packet header by adding the first-order
differences for these fields that are stored in the session context
along with the previous uncompressed header. When the second-order
difference is not zero, the magnitude of the change is usually much
smaller than the full number of bits in the field, so the size can be
reduced by encoding the new first-order difference and transmitting
it rather than the absolute value.
The M bit will be set on the first packet of an audio talkspurt and
the last packet of a video frame. If it were treated as a constant
field such that each change required sending the full RTP header,
this would reduce the compression significantly. Therefore, one bit
in the compressed header will carry the M bit explicitly.
If the packets are flowing through an RTP mixer, most commonly for
audio, then the CSRC list and CC count will also change. However,
the CSRC list will typically remain constant during a talkspurt or
longer, so it need be sent only when it changes.
3.3. The protocol
The compression protocol must maintain a collection of shared
information in a consistent state between the compressor and
decompressor. There is a separate session context for each
IP/UDP/RTP packet stream, as defined by a particular combination of
the IP source and destination addresses, UDP source and destination
ports, and the RTP SSRC field. The number of session contexts to be
maintained MAY be negotiated between the compressor and decompressor.
Each context is identified by an 8- or 16-bit identifier, depending
upon the number of contexts negotiated, so the maximum number is
65536. Both uncompressed and compressed packets MUST carry the
context ID and a 4-bit sequence number used to detect packet loss
between the compressor and decompressor. Each context has its own
separate sequence number space so that a single packet loss need only
invalidate one context.
The shared information in each context consists of the following
items:
o The full IP, UDP and RTP headers, possibly including a CSRC
list, for the last packet sent by the compressor or
reconstructed by the decompressor.
o The first-order difference for the IPv4 ID field, initialized to
1 whenever an uncompressed IP header for this context is
received and updated each time a delta IPv4 ID field is received
in a compressed packet.
o The first-order difference for the RTP timestamp field,
initialized to 0 whenever an uncompressed packet for this
context is received and updated each time a delta RTP timestamp
field is received in a compressed packet.
o The last value of the 4-bit sequence number, which is used to
detect packet loss between the compressor and decompressor.
o The current generation number for non-differential coding of UDP
packets with IPv6 (see [3]). For IPv4, the generation number
may be set to zero if the COMPRESSED_NON_TCP packet type,
defined below, is never used.
o A context-specific delta encoding table (see section 3.3.4) may
optionally be negotiated for each context.
In order to communicate packets in the various uncompressed and
compressed forms, this protocol depends upon the link layer being
able to provide an indication of four new packet formats in addition
to the normal IPv4 and IPv6 packet formats:
FULL_HEADER - communicates the uncompressed IP header plus any
following headers and data to establish the uncompressed header
state in the decompressor for a particular context. The FULL-
HEADER packet also carries the 8- or 16-bit session context
identifier and the 4-bit sequence number to establish
synchronization between the compressor and decompressor. The
format is shown in section 3.3.1.
COMPRESSED_UDP - communicates the IP and UDP headers compressed to
6 or fewer bytes (often 2 if UDP checksums are disabled), followed
by any subsequent headers (possibly RTP) in uncompressed form,
plus data. This packet type is used when there are differences in
the usually constant fields of the (potential) RTP header. The
RTP header includes a potentially changed value of the SSRC field,
so this packet may redefine the session context. The format is
shown in section 3.3.3.
COMPRESSED_RTP - indicates that the RTP header is compressed along
with the IP and UDP headers. The size of this header may still be
just two bytes, or more if differences must be communicated. This
packet type is used when the second-order difference (at least in
the usually constant fields) is zero. It includes delta encodings
for those fields that have changed by other than the expected
amount to establish the first-order differences after an
uncompressed RTP header is sent and whenever they change. The
format is shown in section 3.3.2.
CONTEXT_STATE - indicates a special packet sent from the
decompressor to the compressor to communicate a list of context
IDs for which synchronization has or may have been lost. This
packet is only sent across the point-to-point link so it requires
no IP header. The format is shown in section 3.3.5.
When this compression scheme is used with IPv6 as part of the general
header compression framework specified in [3], another packet type
MAY be used:
COMPRESSED_NON_TCP - communicates the compressed IP and UDP
headers as defined in [3] without differential encoding. If it
were used for IPv4, it would require one or two bytes more than
the COMPRESSED_UDP form listed above in order to carry the IPv4 ID
field. For IPv6, there is no ID field and this non-differential
compression is more resilient to packet loss.
Assignments of numeric codes for these packet formats in the Point-
to-Point Protocol [4] are to be made by the Internet Assigned Numbers
Authority.
3.3.1. FULL_HEADER (uncompressed) packet format
The definition of the FULL_HEADER packet given here is intended to be
the consistent with the definition given in [3]. Full details on
design choices are given there.
The format of the FULL_HEADER packet is the same as that of the
original packet. In the IPv4 case, this is usually an IP header,
followed by a UDP header and UDP payload that may be an RTP header
and its payload. However, the FULL_HEADER packet may also carry IP
encapsulated packets, in which case there would be two IP headers
followed by UDP and possibly RTP. Or in the case of IPv6, the packet
may be built of some combination of IPv6 and IPv4 headers. Each
successive header is indicated by the type field of the previous
header, as usual.
The FULL_HEADER packet differs from the corresponding normal IPv4 or
IPv6 packet in that it must also carry the compression context ID and
the 4-bit sequence number. In order to avoid expanding the size of
the header, these values are inserted into length fields in the IP
and UDP headers since the actual length may be inferred from the
length provided by the link layer. Two 16-bit length fields are
needed; these are taken from the first two available headers in the
packet. That is, for an IPv4/UDP packet, the first length field is
the total length field of the IPv4 header, and the second is the
length field of the UDP header. For an IPv4 encapsulated packet, the
first length field would come from the total length field of the
first IP header, and the second length field would come from the
total length field of the second IP header.
As specified in Sections 5.3.2 of [3], the position of the context ID
(CID) and 4-bit sequence number varies depending upon whether 8- or
16-bit context IDs have been selected, as shown in the following
diagram (16 bits wide, with the most-significant bit is to the left):
For 8-bit context ID:
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0|1| Generation| CID | First length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 0 | seq | Second length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
For 16-bit context ID:
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1|1| Generation| 0 | seq | First length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| CID | Second length field
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The first bit in the first length field indicates the length of the
CID. The length of the CID MUST either be constant for all contexts
or two additional distinct packet types MUST be provided to
separately indicate COMPRESSED_UDP and COMPRESSED_RTP packet formats
with 8- and 16-bit CIDs. The second bit in the first length field is
1 to indicate that the 4-bit sequence number is present, as is always
the case for this IP/UDP/RTP compression scheme.
The generation field is used with IPv6 for COMPRESSED_NON_TCP packets
as described in [3]. For IPv4-only implementations that do not use
COMPRESSED_NON_TCP packets, the compressor SHOULD set the generation
value to zero. For consistent operation between IPv4 and IPv6, the
generation value is stored in the context when it is received by the
decompressor, and the most recent value is returned in the
CONTEXT_STATE packet.
When a FULL_HEADER packet is received, the complete set of headers is
stored into the context selected by the context ID. The 4-bit
sequence number is also stored in the context, thereby
resynchronizing the decompressor to the compressor.
When COMPRESSED_NON_TCP packets are used, the 4-bit sequence number
is inserted into the "Data Field" of that packet and the D bit is set
as described in Section 6 of [3]. When a COMPRESSED_NON_TCP packet
is received, the generation number is compared to the value stored in
the context. If they are not the same, the context is not up to date
and MUST be refreshed by a FULL_HEADER packet. If the generation
does match, then the compressed IP and UDP header information, the
4-bit sequence number, and the (potential) RTP header are all stored
into the saved context.
The amount of memory required to store the context will vary
depending upon how many encapsulating headers are included in the
FULL_HEADER packet. The compressor and decompressor MAY negotiate a
maximum header size.
3.3.2. COMPRESSED_RTP packet format
When the second-order difference of the RTP header from packet to
packet is zero, the decompressor can reconstruct a packet simply by
adding the stored first-order differences to the stored uncompressed
header representing the previous packet. All that need be
communicated is the session context identifier and a small sequence
number (not related to the RTP sequence number) to maintain
synchronization and detect packet loss between the compressor and
decompressor.
If the second-order difference of the RTP header is not zero for some
fields, the new first-order difference for just those fields is
communicated using a compact encoding. The new first-order
difference values are added to the corresponding fields in the
uncompressed header in the decompressor's session context, and are
also stored explicitly in the context to be added to the
corresponding fields again on each subsequent packet in which the
second-order difference is zero. Each time the first-order
difference changes, it is transmitted and stored in the context.
In practice, the only fields for which it is useful to store the
first-order difference are the IPv4 ID field and the RTP timestamp.
For the RTP sequence number field, the usual increment is 1. If the
sequence number changes by other than 1, the difference must be
communicated but does not set the expected difference for the next
packet. Instead, the expected first-order difference remains fixed
at 1 so that the difference need not be explicitly communicated on
the next packet assuming it is in order.
For the RTP timestamp, when a FULL_HEADER, COMPRESSED_NON_TCP or
COMPRESSED_UDP packet is sent to refresh the RTP state, the stored
first-order difference is initialized to zero. If the timestamp is
the same on the next packet (e.g., same video frame), then the
second-order difference is zero. Otherwise, the difference between
the timestamps of the two packets is transmitted as the new first-
order difference to be added to the timestamp in the uncompressed
header stored in the decompressor's context and also stored as the
first-order difference in that context. Each time the first-order
difference changes on subsequent packets, that difference is again
transmitted and used to update the context.
Similarly, since the IPv4 ID field frequently increments by one, the
first-order difference for that field is initialized to one when the
state is refreshed by a FULL_HEADER packet, or when a
COMPRESSED_NON_TCP packet is sent since it carries the ID field in
uncompressed form. Thereafter, whenever the first-order difference
changes, it is transmitted and stored in the context.
A bit mask will be used to indicate which fields have changed by
other than the expected difference. In addition to the small link
sequence number, the list of items to be conditionally communicated
in the compressed IP/UDP/RTP header is as follows:
I = IPv4 packet ID (always 0 if no IPv4 header)
U = UDP checksum
M = RTP marker bit
S = RTP sequence number
T = RTP timestamp
L = RTP CSRC count and list
If 4 bits are needed for the link sequence number to get a reasonable
probability of loss detection, there are too few bits remaining to
assign one bit to each of these items and still fit them all into a
single byte to go along with the context ID.
It is not necessary to explicitly carry the U bit to indicate the
presence of the UDP checksum because a source will typically include
check-sums on all packets of a session or none of them. When the
session state is initialized with an uncompressed header, if there is
a nonzero checksum present, an unencoded 16-bit checksum will be
inserted into the compressed header in all subsequent packets until
this setting is changed by sending another uncompressed packet.
Of the remaining items, the L bit for the CSRC count and list may be
the one least frequently used. Rather than dedicating a bit in the
mask to indicate CSRC change, an unusual combination of the other
bits may be used instead. This bit combination is denoted MSTI. If
all four of the bits for the IP packet ID, RTP marker bit, RTP
sequence number and RTP timestamp are set, this is a special case
indicating an extended form of the compressed RTP header will follow.
That header will include an additional byte containing the real
values of the four bits plus the CC count. The CSRC list, of length
indicated by the CC count, will be included just as it appears in the
uncompressed RTP header.
The other fields of the RTP header (version, P bit, X bit, payload
type and SSRC identifier) are assumed to remain relatively constant.
In particular, the SSRC identifier is defined to be constant for a
given context because it is one of the factors selecting the context.
If any of the other fields change, the uncompressed RTP header MUST
sent as described in Section 3.3.3.
The following diagram shows the compressed IP/UDP/RTP header with
dotted lines indicating fields that are conditionally present. The
most significant bit is numbered 0. Multi-byte fields are sent in
network byte order (most significant byte first). The delta fields
are often a single byte as shown but may be two or three bytes
depending upon the delta value as explained in Section 3.3.4.
0 1 2 3 4 5 6 7
+...............................+
: msb of session context ID : (if 16-bit CID)
+-------------------------------+
| lsb of session context ID |
+---+---+---+---+---+---+---+---+
| M | S | T | I | link sequence |
+---+---+---+---+---+---+---+---+
: :
+ UDP checksum + (if nonzero in context)
: :
+...............................+
: :
+ "RANDOM" fields + (if encapsulated)
: :
+...............................+
: M'| S'| T'| I'| CC : (if MSTI = 1111)
+...............................+
: delta IPv4 ID : (if I or I' = 1)
+...............................+
: delta RTP sequence : (if S or S' = 1)
+...............................+
: delta RTP timestamp : (if T or T' = 1)
+...............................+
: :
: CSRC list : (if MSTI = 1111
: : and CC nonzero)
: :
+...............................+
: :
: RTP header extension : (if X set in context)
: :
: :
+-------------------------------+
| |
| RTP data |
/ /
/ /
| |
+-------------------------------+
: padding : (if P set in context)
+...............................+
When more than one IPv4 header is present in the context as
initialized by the FULL_HEADER packet, then the IP ID fields of
encapsulating headers MUST be sent as absolute values as described in
[3]. These fields are identified as "RANDOM" fields. They are
inserted into the COMPRESSED_RTP packet in the same order as they
appear in the original headers, immediately following the UDP
checksum if present or the MSTI byte if not, as shown in the diagram.
Only if an IPv4 packet immediately precedes the UDP header will the
IP ID of that header be sent differentially, i.e., potentially with
no bits if the second difference is zero, or as a delta IPv4 ID field
if not. If there is not an IPv4 header immediately preceding the UDP
header, then the I bit MUST be 0 and no delta IPv4 ID field will be
present.
3.3.3. COMPRESSED_UDP packet format
If there is a change in any of the fields of the RTP header that are
normally constant (such as the payload type field), then an
uncompressed RTP header MUST be sent. If the IP and UDP headers do
not also require updating, this RTP header MAY be carried in a
COMPRESSED_UDP packet rather than a FULL_HEADER packet. The
COMPRESSED_UDP packet has the same format as the COMPRESSED_RTP
packet except that the M, S and T bits are always 0 and the
corresponding delta fields are never included:
0 1 2 3 4 5 6 7
+...............................+
: msb of session context ID : (if 16-bit CID)
+-------------------------------+
| lsb of session context ID |
+---+---+---+---+---+---+---+---+
| 0 | 0 | 0 | I | link sequence |
+---+---+---+---+---+---+---+---+
: :
+ UDP checksum + (if nonzero in context)
: :
+...............................+
: :
+ "RANDOM" fields + (if encapsulated)
: :
+...............................+
: delta IPv4 ID : (if I = 1)
+-------------------------------+
| UDP data |
: (uncompressed RTP header) :
Note that this constitutes a form of IP/UDP header compression
different from COMPRESSED_NON_TCP packet type defined in [3]. The
motivation is to allow reaching the target of two bytes when UDP
checksums are disabled, as IPv4 allows. The protocol in [3] does not
use differential coding for UDP packets, so in the IPv4 case, two
bytes of IP ID, and two bytes of UDP checksum if nonzero, would
always be transmitted in addition to two bytes of compression prefix.
For IPv6, the COMPRESSED_NON_TCP packet type MAY be used instead.
3.3.4. Encoding of differences
The delta fields in the COMPRESSED_RTP and COMPRESSED_UDP packets are
encoded with a variable-length mapping for compactness of the more
commonly-used values. A default encoding is specified below, but it
is RECOMMENDED that implementations use a table-driven delta encoder
and decoder to allow negotiation of a table specific for each session
if appropriate, possibly even an optimal Huffman encoding. Encodings
based on sequential interpretation of the bit stream, of which this
default table and Huffman encoding are examples, allow a reasonable
table size and may result in an execution speed faster than a non-
table-driven implementation with explicit tests for ranges of values.
The default delta encoding is specified in the following table. This
encoding was designed to efficiently encode the small changes that
may occur in the IP ID and in RTP sequence number when packets are
lost upstream from the compressor, yet still handling most audio and
video deltas in two bytes. The column on the left is the decimal
value to be encoded, and the column on the right is the resulting
sequence of bytes shown in hexadecimal and in the order in which they
are transmitted (network byte order). The first and last values in
each contiguous range are shown, with ellipses in between:
Decimal Hex
-16384 C0 00 00
: :
-129 C0 3F 7F
-128 80 00
: :
-1 80 7F
0 00
: :
127 7F
128 80 80
: :
16383 BF FF
16384 C0 40 00
: :
4194303 FF FF FF
For positive values, a change of zero through 127 is represented
directly in one byte. If the most significant two bits of the byte
are 10 or 11, this signals an extension to a two- or three-byte
value, respectively. The least significant six bits of the first
byte are combined, in decreasing order of significance, with the next
one or two bytes to form a 14- or 22-bit value.
Negative deltas may occur when packets are misordered or in the
intentionally out-of-order RTP timestamps on MPEG video [5]. These
events are less likely, so a smaller range of negative values is
encoded using otherwise redundant portions of the positive part of
the table.
A change in the RTP timestamp value less than -16384 or greater than
4194303 forces the RTP header to be sent uncompressed using a
FULL_HEADER, COMPRESSED_NON_TCP or COMPRESSED_UDP packet type. The
IP ID and RTP sequence number fields are only 16 bits, so negative
deltas for those fields SHOULD be masked to 16 bits and then encoded
(as large positive 16-bit numbers).
3.3.5. Error Recovery
Whenever the 4-bit sequence number for a particular context
increments by other than 1, except when set by a FULL_HEADER or
COMPRESSED_NON_TCP packet, the decompressor MUST invalidate that
context and send a CONTEXT_STATE packet back to the compressor
indicating that the context has been invalidated. All packets for
the invalid context MUST be discarded until a FULL_HEADER or
COMPRESSED_NON_TCP packet is received for that context to re-
establish consistent state (unless the "twice" algorithm is used as
described later in this section). Since multiple compressed packets
may arrive in the interim, the decompressor SHOULD NOT retransmit the
CONTEXT_STATE packet for every compressed packet received, but
instead SHOULD limit the rate of retransmission to avoid flooding the
reverse channel.
When an error occurs on the link, the link layer will usually discard
the packet that was damaged (if any), but may provide an indication
of the error. Some time may elapse before another packet is
delivered for the same context, and then that packet would have to be
discarded by the decompressor when it is observed to be out of
sequence, resulting in at least two packets lost. To allow faster
recovery if the link does provide an explicit error indication, the
decompressor MAY optionally send an advisory CONTEXT_STATE packet
listing the last valid sequence number and generation number for one
or more recently active contexts (not necessarily all). For a given
context, if the compressor has sent no compressed packet with a
higher sequence number, and if the generation number matches the
current generation, no corrective action is required. Otherwise, the
compressor MAY choose to mark the context invalid so that the next
packet is sent in FULL_HEADER or COMPRESSED_NON_TCP mode (FULL_HEADER
is required if the generation doesn't match). However, note that if
the link round-trip-time is large compared to the inter-packet
spacing, there may be several packets from multiple contexts in
flight across the link, increasing the probability that the sequence
numbers will already have advanced when the CONTEXT_STATE packet is
received by the compressor. The result could be that some contexts
are invalidated unnecessarily, causing extra bandwidth to be
consumed.
The format of the CONTEXT_STATE packet is shown in the following
diagrams. The first byte is a type code to allow the CONTEXT_STATE
packet type to be shared by multiple compression schemes within the
general compression framework specified in [3]. The contents of the
remainder of the packet depends upon the compression scheme. For the
IP/UDP/RTP compression scheme specified here, the remainder of the
CONTEXT_STATE packet is structured as a list of blocks to allow the
state for multiple contexts to be indicated, preceded by a one-byte
count of the number of blocks.
Two type code values are used for the IP/UDP/RTP compression scheme.
The value 1 indicates that 8-bit session context IDs are being used:
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
| 1 = IP/UDP/RTP with 8-bit CID |
+---+---+---+---+---+---+---+---+
| context count |
+---+---+---+---+---+---+---+---+
+---+---+---+---+---+---+---+---+
| session context ID |
+---+---+---+---+---+---+---+---+
| I | 0 | 0 | 0 | sequence |
+---+---+---+---+---+---+---+---+
| 0 | 0 | generation |
+---+---+---+---+---+---+---+---+
...
+---+---+---+---+---+---+---+---+
| session context ID |
+---+---+---+---+---+---+---+---+
| I | 0 | 0 | 0 | sequence |
+---+---+---+---+---+---+---+---+
| 0 | 0 | generation |
+---+---+---+---+---+---+---+---+
The value 2 indicates that 16-bit session context IDs are being used.
The session context ID is sent in network byte order (most
significant byte first):
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
| 2 = IP/UDP/RTP with 16-bit CID|
+---+---+---+---+---+---+---+---+
| context count |
+---+---+---+---+---+---+---+---+
+---+---+---+---+---+---+---+---+
| |
+ session context ID +
| |
+---+---+---+---+---+---+---+---+
| I | 0 | 0 | 0 | sequence |
+---+---+---+---+---+---+---+---+
| 0 | 0 | generation |
+---+---+---+---+---+---+---+---+
...
+---+---+---+---+---+---+---+---+
| |
+ session context ID +
| |
+---+---+---+---+---+---+---+---+
| I | 0 | 0 | 0 | sequence |
+---+---+---+---+---+---+---+---+
| 0 | 0 | generation |
+---+---+---+---+---+---+---+---+
The bit labeled "I" is set to one for contexts that have been marked
invalid and require a FULL_HEADER of COMPRESSED_NON_TCP packet to be
transmitted. If the I bit is zero, the context state is advisory.
The I bit is set to zero to indicate advisory context state that MAY
be sent following a link error indication.
Since the CONTEXT_STATE packet itself may be lost, retransmission of
one or more blocks is allowed. It is expected that retransmission
will be triggered only by receipt of another packet, but if the line
is near idle, retransmission MAY be triggered by a relatively long
timer (on the order of 1 second).
If a CONTEXT_STATE block for a given context is retransmitted, it may
cross paths with the FULL_HEADER or COMPRESSED_NON_TCP packet
intended to refresh that context. In that case, the compressor MAY
choose to ignore the error indication.
In the case where UDP checksums are being transmitted, the
decompressor MAY attempt to use the "twice" algorithm described in
section 10.1 of [3]. In this algorithm, the delta is applied more
than once on the assumption that the delta may have been the same on
the missing packet(s) and the one subsequently received. Success is
indicated by a checksum match. For the scheme defined here, the
difference in the 4- bit sequence number tells number of times the
delta must be applied. Note, however, that there is a nontrivial
risk of an incorrect positive indication. It may be advisable to
request a FULL_HEADER or COMPRESSED_NON_TCP packet even if the
"twice" algorithm succeeds.
Some errors may not be detected, for example if 16 packets are lost
in a row and the link level does not provide an error indication. In
that case, the decompressor will generate packets that are not valid.
If UDP checksums are being transmitted, the receiver will probably
detect the invalid packets and discard them, but the receiver does
not have any means to signal the decompressor. Therefore, it is
RECOMMENDED that the decompressor verify the UDP checksum
periodically, perhaps one out of 16 packets. If an error is
detected, the decompressor would invalidate the context and signal
the compressor with a CONTEXT_STATE packet.
3.4. Compression of RTCP Control Packets
By relying on the RTP convention that data is carried on an even port
number and the corresponding RTCP packets are carried on the next
higher (odd) port number, one could tailor separate compression
schemes to be applied to RTP and RTCP packets. For RTCP, the
compression could apply not only to the header but also the "data",
that is, the contents of the different packet types. The numbers in
Sender Report (SR) and Receiver Report (RR) RTCP packets would not
compress well, but the text information in the Source Description
(SDES) packets could be compressed down to a bit mask indicating each
item that was present but compressed out (for timing purposes on the
SDES NOTE item and to allow the end system to measure the average
RTCP packet size for the interval calculation).
However, in the compression scheme defined here, no compression will
be done on the RTCP headers and "data" for several reasons (though
compression SHOULD still be applied to the IP and UDP headers).
Since the RTP protocol specification suggests that the RTCP packet
interval be scaled so that the aggregate RTCP bandwidth used by all
participants in a session will be no more than 5% of the session
bandwidth, there is not much to be gained from RTCP compression.
Compressing out the SDES items would require a significant increase
in the shared state that must be stored for each context ID. And, in
order to allow compression when SDES information for several sources
was sent through an RTP "mixer", it would be necessary to maintain a
separate RTCP session context for each SSRC identifier. In a session
with more than 255 participants, this would cause perfect thrashing
of the context cache even when only one participant was sending data.
Even though RTCP is not compressed, the fraction of the total
bandwidth occupied by RTCP packets on the compressed link remains no
more than 5% in most cases, assuming that the RTCP packets are sent
as COMPRESSED_UDP packets. Given that the uncompressed RTCP traffic
consumes no more than 5% of the total session bandwidth, then for a
typical RTCP packet length of 90 bytes, the portion of the compressed
bandwidth used by RTCP will be no more than 5% if the size of the
payload in RTP data packets is at least 108 bytes. If the size of
the RTP data payload is smaller, the fraction will increase, but is
still less than 7% for a payload size of 37 bytes. For large data
payloads, the compressed RTCP fraction is less than the uncompressed
RTCP fraction (for example, 4% at 1000 bytes).
3.5. Compression of non-RTP UDP Packets
As described earlier, the COMPRESSED_UDP packet MAY be used to
compress UDP packets that don't carry RTP. Whatever data follows the
UDP header is unlikely to have some constant values in the bits that
correspond to usually constant fields in the RTP header. In
particular, the SSRC field would likely change. Therefore, it is
necessary to keep track of the non-RTP UDP packet streams to avoid
using up all the context slots as the "SSRC field" changes (since
that field is part of what identifies a particular RTP context).
Those streams may each be given a context, but the encoder would set
a flag in the context to indicate that the changing SSRC field should
be ignored and COMPRESSED_UDP packets should always be sent instead
of COMPRESSED_RTP packets.
4. Interaction With Segmentation
A segmentation scheme may be used in conjunction with RTP header
compression to allow small, real-time packets to interrupt large,
presumably non-real-time packets in order to reduce delay. It is
assumed that the large packets bypass the compressor and decompressor
since the interleaving would modify the sequencing of packets at the
decompressor and cause the appearance of errors. Header compression
should be less important for large packets since the overhead ratio
is smaller.
If some packets from an RTP session context are selected for
segmentation (perhaps based on size) and some are not, there is a
possibility of re-ordering. This would reduce the compression
efficiency because the large packets would appear as lost packets in
the sequence space. However, this should not cause more serious
problems because the RTP sequence numbers should be reconstructed
correctly and will allow the application to correct the ordering.
Link errors detected by the segmentation scheme using its own
sequencing information MAY be indicated to the compressor with an
advisory CONTEXT_STATE message just as for link errors detected by
the link layer itself.
The context ID byte is placed first in the COMPRESSED_RTP header so
that this byte MAY be shared with the segmentation layer if such
sharing is feasible and has been negotiated. Since the compressor
may assign context ID values arbitrarily, the value can be set to
match the context identifier from the segmentation layer.
5. Negotiating Compression
The use of IP/UDP/RTP compression over a particular link is a
function of the link-layer protocol. It is expected that such
negotiation will be defined separately for PPP [4], for example. The
following items MAY be negotiated:
o The size of the context ID.
o The maximum size of the stack of headers in the context.
o A context-specific table for decoding of delta values.
6. Acknowledgments
Several people have contributed to the design of this compression
scheme and related problems. Scott Petrack initiated discussion of
RTP header compression in the AVT working group at Los Angeles in
March, 1996. Carsten Bormann has developed an overall architecture
for compression in combination with traffic control across a low-
speed link, and made several specific contributions to the scheme
described here. David Oran independently developed a note based on
similar ideas, and suggested the use of PPP Multilink protocol for
segmentation. Mikael Degermark has contributed advice on integration
of this compression scheme with the IPv6 compression framework.
7. References:
[1] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:
A Transport Protocol for real-time applications", RFC 1889,
January 1996.
[2] Jacobson, V., "TCP/IP Compression for Low-Speed Serial Links",
RFC 1144, February 1990.
[3] Degermark, M., Nordgren, B. and S. Pink, "Header Compression for
IPv6", RFC 2507, February 1999.
[4] Simpson, W., "The Point-to-Point Protocol (PPP)", STD 51, RFC
1661, July 1994.
[5] Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, "RTP
Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998.
8. Security Considerations
Because encryption eliminates the redundancy that this compression
scheme tries to exploit, there is some inducement to forego
encryption in order to achieve operation over a low-bandwidth link.
However, for those cases where encryption of data and not headers is
satisfactory, RTP does specify an alternative encryption method in
which only the RTP payload is encrypted and the headers are left in
the clear. That would allow compression to still be applied.
A malfunctioning or malicious compressor could cause the decompressor
to reconstitute packets that do not match the original packets but
still have valid IP, UDP and RTP headers and possibly even valid UDP
check-sums. Such corruption may be detected with end-to-end
authentication and integrity mechanisms which will not be affected by
the compression. Constant portions of authentication headers will be
compressed as described in [3].
No authentication is performed on the CONTEXT_STATE control packet
sent by this protocol. An attacker with access to the link between
the decompressor and compressor could inject false CONTEXT_STATE
packets and cause compression efficiency to be reduced, probably
resulting in congestion on the link. However, an attacker with
access to the link could also disrupt the traffic in many other ways.
A potential denial-of-service threat exists when using compression
techniques that have non-uniform receiver-end computational load.
The attacker can inject pathological datagrams into the stream which
are complex to decompress and cause the receiver to be overloaded and
degrading processing of other streams. However, this compression
does not exhibit any significant non-uniformity.
A security review of this protocol found no additional security
considerations.
9. Authors' Addresses
Stephen L. Casner
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
United States
EMail: casner@cisco.com
Van Jacobson
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
United States
EMail: van@cisco.com
10. Full Copyright Statement
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or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
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included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
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